1. Field of the Invention
The present invention relates to electronic hearing aid devices for use by the hearing impaired and to methods for providing hearing compensation. More particularly, the present invention relates to using differential signal sampling for digital signal processing in such devices and methods.
2. The Prior Art
In conventional hearing aid systems, a hearing aid typically includes an input transducer, a signal processing circuit, and an output transducer. Acoustical energy detected by the input transducer is changed into an electrical signal that is representative of the acoustical energy. To compensate for the hearing deficiencies of the hearing aid user, the signal processing circuit modifies the electrical signal. The signal processing may occur in a single frequency band or in multiple frequency bands and may be either linear or non-linear. The output transducer transduces the processed signal back into acoustical energy for detection by the ear of the hearing aid user.
One manner known in the art to perform the signal processing in the hearing aid is digital signal processing (DSP). Since the output from the input transducer is typically an analog electrical signal, the analog electrical signal is converted to a digital signal by an analog-to-digital (A/D) converter. The precision with which the DSP operations are performed depends generally on two things. First, the precision of the operations themselves, and second, the number of digital bits being output from the A/D converter to represent each digital sample of the signal being fed into the DSP operations. Accordingly, more bits are used to increase the precision of the sample and the accuracy with which the signal can be processed.
In conventional hearing aid systems which employ DSP techniques, the A/D conversion may be implemented using any one of a number of general A/D converters including flash or parallel converters, iterative converters, ramp or staircase converters, tracking converters, integrating converters, and sigma-delta converters followed by an integrator.
The DSP operations are performed on the digital output of the A/D converter representing the full magnitude of the analog input signal. While seeking to have an adequate number of bits for accurate DSP operations, using the smallest number of bits has important advantages. A first advantage is that with fewer bits to process, the energy consumption of the circuits performing the input, output, and the modification of the signal is reduced. A second advantage is that the complexity of the circuits performing the input, the output and the modification of the signal is also reduced. In a hearing aid system, minimizing both the size of the device and the power consumption of the device are important objectives.
It has also been recognized that in individuals with hearing loss, the degree of hearing loss may not be the same across the entire audio spectrum. Accordingly, the audio signal in different frequency bands is digital signal processed in each separate frequency band according to parameters selected to compensate for the hearing loss in that particular frequency band.
The DSP in each frequency band may be either linear or non-linear, however, when the DSP is non-linear, a problem not encountered in linear systems must be addressed. In linear systems, a signal which has been split into several different frequency bands and then linearly digitally processed in each frequency band is summed back together after the DSP according to the law of Linear Superposition.
For non-linear systems it is known that there is no generalized law of Superposition. One approach to providing a rule for superposition in non-linear systems has been set forth by Oppenheim et al, in Nonlinear Filtering of Multiplied and Convoluted Signals. Proc. IEE, Vol. 56, pp. 1264-1291, August 1968, which proposed a generalized law of superposition for a class of non-linear systems which can be treated as linear after a transformation. This class of non-linear systems are referred to as homomorphic systems. An example of a homomorphic system can be found in U.S. Pat. No. 5,500,902, wherein a logarithm of the input signal in each frequency band is first taken before additional signal processing is performed on the input signal. The antilog of the processed signal is then taken, and the signals from each frequency band are summed.
It is therefore an object of the present invention to minimize the number of bits in the sampled digital representation of the signal being processed by the hearing aid system.
It is another object of the present invention to minimize the number of bits in the sampled digital representation of the signal by representing the difference between successive analog input signal samples as the sampled digital signal.
It is yet another object of the present invention to use a differential digital signal sample as the digital signal in a multiband hearing aid system.
It is a further object of the present invention to use a differential digital signal sample as the digital signal in a multiband sound processing system.
It is therefore an object of the present invention to implement a multiband hearing aid using a homomorphic transformation in the DSP operations with a differential signal sample representation.
It is another object of the present invention to implement a multiband hearing aid using non-linear DSP operations with differential signal sample representation.
It is a further object of the present invention to implement a multiband sound processing system using non-linear DSP operations with differential signal sample representation.
It is a further object of the present invention to implement a multiband hearing aid using a table look-up for DSP operations with differential signal sample representation.
It is yet another object of the present invention to implement a multiband sound processing system using a table look-up for DSP operations with differential signal sample representation.
It is a further object of the present invention to implement a sound processing system wherein the output transducer is driven by pulses having widths proportional to a differential digital signal.